Issues 25000 - 25999

[..]
ASTERISK-25000: Deadlock in ast_do_masquerade (specifically in ast_hangup on the zombie clone if it's hungup during the masquerade)
ASTERISK-25001: SendFAX error
ASTERISK-25002: [patch] - Demote "Extension Changed" messages in Asterisk CLI
ASTERISK-25003: Asterisk crashes on attended transfer (using feature)
ASTERISK-25004: Crash in authenticated reinvite after originated T.38 FAX
ASTERISK-25005: MixMonitor doesn't record outgoing calls
ASTERISK-25006: [patch] Add support set character for quoted identifiers
ASTERISK-25007: Notify packet to private IP endpoint behind nat with pjsip tls transport
ASTERISK-25008: While Redirecting dual channel and one of them is AGENT channel. Agent loggs off form the system
ASTERISK-25009: DNS Tests: SRV Priority
ASTERISK-25010: DNS Tests: Failover order
ASTERISK-25011: DNS Tests: Failover to A/AAAA
ASTERISK-25012: DNS Tests: NAPTR Nominal - Correct Order
ASTERISK-25013: DNS Tests: NAPTR Nominal - Correct Preference
ASTERISK-25014: DNS Tests: NAPTR Nominal - Restricted Transport
ASTERISK-25015: DNS Tests: NAPTR Nominal - Failover of preferences
ASTERISK-25016: one-sided audio when auto_comedia is set
ASTERISK-25017: download from svn fails
ASTERISK-25018: pjsip show endpoints crashes asterisk when qualified aors present
ASTERISK-25019: called party Unhold SRTP calls cause NO MEDIA. unprotect failed with: authentication failure 110
ASTERISK-25020: Mismatched response to outgoing REGISTER request
ASTERISK-25021: Fix invalid pointer dereference on module load
ASTERISK-25022: Memory leak setting up DTLS/SRTP calls
ASTERISK-25023: Deadlock in chan_sip in update_provisional_keepalive
ASTERISK-25024: there is no ice when i make originale
ASTERISK-25025: Periodic crashes (in ast_channel_snapshot_create at stasis_channels.c) with Certified Asterisk 13.
ASTERISK-25026: Git conversion: Non-C files not switched to ASTERISK_REGISTER_FILE
ASTERISK-25027: Build System: Many ARI modules are missing dependencies.
ASTERISK-25028: Build System: Unneeded defines in asterisk/buildopts.h
ASTERISK-25029: Astobj2: Create ao2_weakproxy_ref_object function.
ASTERISK-25030: Avoid using LIKE when using realtime engine
ASTERISK-25031: DTMF INFO before answer leads to 200 OK without Contact: header
ASTERISK-25032: [patch]cel_odbc sometimes inserts CEL with wrong eventtime
ASTERISK-25033: Asterisk 13 (branch head) won't compile without PJSip
ASTERISK-25034: chan_dahdi: Some telco switches occasionally ignore ISDN RESTART requests.
ASTERISK-25035: Stuck Channels
ASTERISK-25036: Asterisk still send an INVITE request after a call was canceled(realtime, rtcachefriends is enabled)
ASTERISK-25037: res_pjsip_outbound_registration: Potential crash in off-nominal failure case when sending message
ASTERISK-25038: Queue log "EXITWITHTIMEOUT" does not always contain waiting time
ASTERISK-25039: getting major delays when connecting a call to a webrtc client
ASTERISK-25040: pbx: Improve performance of reloads by making hint destruction more performant
ASTERISK-25041: [patch]Broken column type checking in res_config_mysql addon
ASTERISK-25042: asterisk.conf options override command-line options.
ASTERISK-25043: [patch] Avoiding ERR_remove_state in OpenSSL
ASTERISK-25044: sorcery: Add ability to insert a new wizard into an object type's list
ASTERISK-25045: vector: Add new capabilities and unit tests
ASTERISK-25046: Chan_sip deadlock persistent between Asterisk 1.8 versions, using Realtime.
ASTERISK-25047: Call remaining in "core show channels" after hangup
ASTERISK-25048: Astobj2: Initialization order wrong when both refdebug and AO2_DEBUG are both enabled.
ASTERISK-25049: CLI: Enable automatic references to modules
ASTERISK-25050: res_pjsip cannot load when contact is enabled for realtime
ASTERISK-25051: Remove unneeded uses of optional_api providers.
ASTERISK-25052: OPTIONAL_API: Remove ABI option.
ASTERISK-25053: Unit test category /main/presence missing trailing slash.
ASTERISK-25054: Formats interface's cannot be unregistered, needs to hold modules until shutdown.
ASTERISK-25055: ARI Snoop Channel: Improve documentation
ASTERISK-25056: Modules: Make ast_module_info->self available to auxiliary sources.
ASTERISK-25057: res_pjsip_pubsub: Crash in send_notify due to invalid root pointer in sub_tree
ASTERISK-25058: res_pjsip_pubsub: Crash in ast_sorcery_hash called from subscription_persistence_update
ASTERISK-25059: recording calls in ogg format creates memory leak
ASTERISK-25060: configure - libxml2 not found
ASTERISK-25061: pbx_config: Register manager actions with module version of macro.
ASTERISK-25063: [patch]add X.509 subject alternative name support to Asterisk TLS support
ASTERISK-25064: Members (ringinuse disabled) of multiple queues ringing with other queue calls.
ASTERISK-25065: SRTP failing over time
ASTERISK-25066: DTMFs are not sent to the bridge channel if they are used by any built-in or dynamic feature
ASTERISK-25067: Sorcery Caching: Implement a new caching module
ASTERISK-25068: Move commonly used FreePBX extra sounds to the core set
ASTERISK-25069: main/message.c: Fix unregister functions
ASTERISK-25070: Fix FTBFS on Hurd
ASTERISK-25071: RFC3581 compliance
ASTERISK-25072: res_pjsip_outbound_registration: line functionality. Additional check for using the request URI
ASTERISK-25073: Asterisk crash by ast_format_cap_append
ASTERISK-25074: Regression: Recent clang-related change broke cross compiling of Asterisk
ASTERISK-25075: func_speex.c does not compile, missing speex/speex_preprocess.h
ASTERISK-25076: res_pjsip: Failover does not occur on connection-less transport or 503 response
ASTERISK-25077: use IP_FREEBIND on network sockets
ASTERISK-25078: sig_pri: Publish progress codes.
ASTERISK-25079: AMI bridge of channels results in MOH not destroyed and robotic audio on one channel
ASTERISK-25080: res_pjsip_refer: Refer code invoked with unexpected NULL channel on session
ASTERISK-25081: res_pjsip_refer: Refer code invoked with unexpected NULL channel on session
ASTERISK-25082: Asterisk deletes message after doing a playback of an INBOX message using ast_vm_play when the Old folder is full for that mailbox.
ASTERISK-25083: Message.c: Message channel becomes saturated with frames leading to spammy log messages
ASTERISK-25084: silenceSupp is not sent by default
ASTERISK-25085: [patch]Potential crash after unload of func_periodic_hook or test_message
ASTERISK-25086: [patch]PJSIP crashes if endpoint missing in Dial()
ASTERISK-25087: Asterisk segfault when using Directory application with alias option and specific mailbox configuration
ASTERISK-25088: res_pjsip: Failure to specify cipher in a TLS transport causes a SIGABRT in pjproject
ASTERISK-25089: res_pjsip_config_wizard: Variable specified in templates aren't being processed correctly
ASTERISK-25090: CLI core show channel truncates cdr variables
ASTERISK-25091: Asterisk REST API - bridge.addChannel crash asterisk when calling channel hangup while adding to bridge
ASTERISK-25092: xmpp connection failure causes continuous error messages
ASTERISK-25093: Asterisk stop working suddenly often
ASTERISK-25094: PBX core: Investigate thread safety issues
ASTERISK-25095: Not manage "busy/congestion" in Asterisk release after v. 1.8.29.0
ASTERISK-25096: [patch]Segfault when registering over websockets with PJSIP (in ast_sockaddr_isnull at /include/asterisk/netsock2.h)
ASTERISK-25097: Asterisk13.3.2 PJSIP configuration user_agent doesn't go into effect
ASTERISK-25098: Auto-close for JIRA issues with merged commits is not functioning
ASTERISK-25099: res_rtp_asterisk: Crash when using DTLS
ASTERISK-25100: asterisk coredump if host has an IPv6 address that end with ::80
ASTERISK-25101: DTLS configuration can not be specified in the general section - documentation
ASTERISK-25102: res_config_odbc / libodbc - Asterisk core dumps with signal 6 when connection lost during query
ASTERISK-25103: Roundup - investigate Asterisk DTLS crashes
ASTERISK-25104: Unnecessary Unlink event on reINVITE when using Monitor()
ASTERISK-25105: res_pjsip: Possible incompatibility between qualify_timeout and pjproject-2.4
ASTERISK-25106: Segmentation fault in ast_variable_new when using app_voicemail with realtime
ASTERISK-25107: Phone losing registration 10 seconds after reboot.
ASTERISK-25108: configure check for older unbound library
ASTERISK-25109: [patch] CEL and CDR - Assigned separator for column name and values.
ASTERISK-25110: res_resolver_unbound.c compilation failure: SIGURG is undeclared in func unbound_resolver_stop
ASTERISK-25111: Playback fails when matching h264 codec
ASTERISK-25112: Logger: Configuration settings are not reset to default during reload.
ASTERISK-25113: install_prereq in Debian 8 without "standard system utilities"
ASTERISK-25114: res_pjsip: Add AMI events for chan_pjsip contact lifecycle changes
ASTERISK-25115: Crash related to func sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c
ASTERISK-25116: res_pjsip: Two PeerStatus AMI messages are sent for every status change
ASTERISK-25117: res_mwi_external_ami: Fix manager action registrations.
ASTERISK-25118: [patch]PeerStatus Event Unsubscribed with Peer IP Address
ASTERISK-25119: Crash on pjsip_tls_transport_start2
ASTERISK-25120: Astobj2: Weakproxy subscriptions should be run in reverse order.
ASTERISK-25121: Stasis: Fix unsafe use of stasis_unsubscribe in modules.
ASTERISK-25122: Large SIP packet received via pjsip over websocket crashes Asterisk
ASTERISK-25123: Bracketed IPv6 Contact header parameter unparsable with Asterisk/PJSIP
ASTERISK-25124: menuselect configure unnecessarily requires libxml2 for 11.18.0-rc1
ASTERISK-25125: Testsuite: Investigate lock-up of tests/fastagi/record-file
ASTERISK-25126: process_sdp: Can't provide secure audio requested in SDP offer
ASTERISK-25127: DTLS crashes following "Unable to cancel schedule ID" in dtls_srtp_check_pending
ASTERISK-25128: Datastore: Implement automatic module references.
ASTERISK-25129: wrong automatic ras address assignment if multihomed
ASTERISK-25130: Load average very high with Local channels
ASTERISK-25131: chan_pjsip: In-dialog authentication not handled.
ASTERISK-25132: escaping manually
ASTERISK-25133: PCI Voice/Data/Fax Modem
ASTERISK-25134: Problem with CDR information with incoming calls
ASTERISK-25135: [patch]RTP Timeout hangup cause code missing
ASTERISK-25136: gosub issue
ASTERISK-25137: endpoint stasis messages are delivered twice
ASTERISK-25138: Unclosed parenthesis in AGI argument leads to further arguments concatenated - parameter quoting not respected
ASTERISK-25139: Malicious transfer sequence locks up Asterisk
ASTERISK-25141: pjsip_options: Contact reference leak
ASTERISK-25142: G722 recordings and spy does not give audio in asterisk => 11.17
ASTERISK-25143: segmentation fault in ast_format_get_type
ASTERISK-25144: Testsuite: Investigate failures in tests for channels/pjsip/subscriptions/rls
ASTERISK-25145: Unable to record calls picked up from parking lot
ASTERISK-25146: DNS: Create system level resolver
ASTERISK-25147: [patch]Testsuite: tests/pjsip/transfers/blind_transfer/caller_with_hold fails after PJSIP 2.4 upgrade
ASTERISK-25148: res_pjsip NULL channel audit
ASTERISK-25149: res_pjsip_session: Attended transfer nominal callee local direct media test failing
ASTERISK-25150: chan_pjsip: RLS subscriptions produce leaks
ASTERISK-25151: results of the automated codereview of the Asterisk project
ASTERISK-25152: func_cdr: Deadlock when used with ARI originated channels
ASTERISK-25153: Asterisk originate with Application/data does not go to dialplan
ASTERISK-25154: [patch]fromtag may need to be updated after successful call dialog match
ASTERISK-25155: Musiconhold bug second time caller is put on hold
ASTERISK-25156: chan_pjsip’s CHAN_START cel event lacks the correct context and exten
ASTERISK-25157: bridging: Performing a blonde transfer does not result in connected line updates
ASTERISK-25158: res_pjsip: Add option to use AAL2 packing when negotiating g.726
ASTERISK-25159: queue_log does not log PAUSE events for pause states set with QUEUE_MEMBER
ASTERISK-25160: [patch] Opus Codec: SIP/SDP line fmtp missing when called internally
ASTERISK-25161: PJSIP "connected_line_method=update" segfault in libpjmedia.so during attended transfer
ASTERISK-25162: func_pjsip_aor: Leak of contact in iterator
ASTERISK-25163: Deadlock in chan_sip between reload of sip peer container and MWI Stasis callback
ASTERISK-25164: asterisk 11.9
ASTERISK-25165: Testsuite - Sorcery memory cache leaks
ASTERISK-25166: No audio when using direct media and a codec with a dynamic payload
ASTERISK-25167: Testsuite: Resolve remaining Asterisk shutdown timeout's
ASTERISK-25168: Random Core Dumps on Asterisk 13.4 PJSIP, in ast_channel_name at channel_internal_api.c
ASTERISK-25169: No audio from voicemail app with v13.4.0 on Grandstream GXP20XX phones
ASTERISK-25170: Segfault in call to vsnprintf from astman_append
ASTERISK-25171: Early completion of feature code attended transfer results in intermittent one-way audio, "ghost ringing" and robotic sound.
ASTERISK-25172: Crash in channels/sip/sip blind transfer/caller_refer_only test in ast_format_cap_append_from_cap during ast_request
ASTERISK-25173: ARI: Add the ability to load/reload/unload an Asterisk module
ASTERISK-25174: Wiki Documentation - Features - Recording (one touch) features
ASTERISK-25175: Wiki Documentation - Features - Application Map & Dynamic Features
ASTERISK-25176: Asterisk segfault on reload res_odbc.so
ASTERISK-25177: Make configure script bail if pjproject was built statically
ASTERISK-25178: Cannot login to Asterisk Forums
ASTERISK-25179: CDR(billsec,f) and CDR(duration,f) report incorrect values
ASTERISK-25180: res_pjsip_mwi: Unsolicited MWI requires reload
ASTERISK-25181: ARI: Channels added to Stasis application during WebSocket creation don't receive a StasisStart event
ASTERISK-25182: [patch] on CLI sip reload, new codecs get appended only
ASTERISK-25183: PJSIP: Crash on NULL channel in chan_pjsip_incoming_response despite previous checks for NULL channel
ASTERISK-25184: Lockup in IAX channel
ASTERISK-25185: Segfault in app_queue on transfer scenarios
ASTERISK-25186: Testing
ASTERISK-25187: Caller or member information missing from app_queue AgentComplete event
ASTERISK-25188: Fax Detection through RFC2833
ASTERISK-25189: AMI: Add Linkedid header to standard channel snapshot information.
ASTERISK-25190: Testing 2
ASTERISK-25191: Testing 3
ASTERISK-25192: Testing 4
ASTERISK-25193: Testing 5
ASTERISK-25194: Incorrect GotoIf Behavoir
ASTERISK-25195: Can I?
ASTERISK-25196: res_pjsip_nat: rewrite_contact should not be applied to Contact header when Record-Route headers are present
ASTERISK-25197: asterisk 13.4.0 fails to transcode calls to/from g729 with TCE400P
ASTERISK-25198: chan_dahdi with SS7 is writing log messages to console despite verbose 0
ASTERISK-25199: Testing Again
ASTERISK-25200: Wanna see something
ASTERISK-25201: Crash in PJSIP distributor on already free'd threadpool
ASTERISK-25202: Hints extension state broken between 13.3.2 and 13.4
ASTERISK-25203: AMI event QueueCallerLeave not always sent
ASTERISK-25204: res_pjsip_refer: Duplicated Referred-By or Replaces headers on outbound INVITEs.
ASTERISK-25205: pjsip Via broken if there are multiple transports with different bind ports
ASTERISK-25206: No ability to control asterisk if permission failure on /var/run
ASTERISK-25207: Yolo
ASTERISK-25208: Testing
ASTERISK-25209: asterisk.org still refers to SVN - update references to SVN in documentation and website
ASTERISK-25210: pjsip 'qualify_timeout' problem
ASTERISK-25211: Prevent timer_pthread from loading with non-compliant systems
ASTERISK-25212: [patch]Segfault when using DEBUG_FD_LEAKS
ASTERISK-25213: [patch]Possibility of deadlock in chan_sip INVITE early Replace code
ASTERISK-25214: DTMF over SIP INFO and direct media does not work well together
ASTERISK-25215: Differences in queue.log between Set QUEUE_MEMBER and using PauseQueueMember
ASTERISK-25216: Asterisk periodic hangs. UDP Recv-Q greatly exceeds zero.
ASTERISK-25217: [patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
ASTERISK-25218: res_config_pgsql freezes asterisk at startup - "init table" request (function find_table) running infinitely
ASTERISK-25219: [patch]Source and destination overlap in memcpy in rtp_engine.c
ASTERISK-25220: [patch]Closing of fd -1 in chan_mgcp.c
ASTERISK-25221: [patch]Improvement to "sip show peer <peer>" - Get additional contact parameters
ASTERISK-25222: Crash in recurring cancel callback called from ast_dns_resolve_cancel on junk pointer
ASTERISK-25223: Crash in RLS MWI termination tests
ASTERISK-25224: WARNING message flooding Asterisk logs
ASTERISK-25225: testsuite: lua tests using SIPp cannot run on Jenkins
ASTERISK-25226: chan_sip: Channel leak in branch 13 on early replaces call pickup
ASTERISK-25227: No audio at in-band announcements in ooh323 channel
ASTERISK-25228: configure with netsnmp fail
ASTERISK-25229: Exchanging Device and Mailbox State Using PJSIP fails after restart of peer
ASTERISK-25230: Crash in channels/pjsip/basic_calls/incoming/off-nominal/userpass when decreasing reference on PJSIP transport
ASTERISK-25231: Set Music AGI command with class argument fails silently - need user level log message
ASTERISK-25232: unistim not showing callerid
ASTERISK-25233: Wiki Documentation - Configuration/Reporting - Call Detail Records
ASTERISK-25234: Wiki Documentation - Configuration/Reporting - Channel Event Logging
ASTERISK-25235: Wiki Documentation - Configuration/Interfaces - Calendaring
ASTERISK-25236: Asterisk reject incoming call from FXO gateway
ASTERISK-25237: stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
ASTERISK-25238: ARI: Support push configuration
ASTERISK-25239: Asterisk 13.4.0 crashes
ASTERISK-25240: bridge_native_rtp: Direct media wrongfully started when completing attended transfer
ASTERISK-25241: Wiki Documentation - Configuration/Interfaces - Asterisk Gateway Interface
ASTERISK-25242: PJSIP: No audio when Asterisk inside NAT and endpoints outside NAT - implement functionality similar to chan_sip 'rtpkeepalive'?
ASTERISK-25243: Asterisk logger can't log level "security" to syslog
ASTERISK-25244: WIki Documentation - Configuration/Interfaces - Asterisk Manager Interface
ASTERISK-25245: Wiki Documentation - Organize and update database connectivity and realtime documentation
ASTERISK-25246: Queues repeatedly try agents in Unavailable status
ASTERISK-25247: choppy audio when spying on a g722 channel, chan_sip or chan_pjsip
ASTERISK-25248: [patch]Improve Chan_Local's bridging speed
ASTERISK-25249: Features code not working for called party when Local channels are involved
ASTERISK-25250: chan_sip - Despite the channel being answered, caller on a call established via Local channel continues to hear ringback
ASTERISK-25251: getifaddrs() blocks infinitely in PJSIP
ASTERISK-25252: ARI: Add the ability to manipulate log channels
ASTERISK-25253: confbridge volume options and other volume controls such as func_volume don't work
ASTERISK-25254: Crash if dialplan sets ATTENDEDTRANSFER to an empty string before Park.
ASTERISK-25255: Missing AMI VarSet events when setting to an empty string.
ASTERISK-25256: [patch]Post AMI VarSet to empty string events when Asterisk deletes a dialplan variable.
ASTERISK-25257: [patch]channels/sig_pri.h -> sig_pri_span -> force_restart_unavailable_chans in wrong scope
ASTERISK-25258: chan_pjsip: Incorrect format switch on received RTP packet
ASTERISK-25259: chan_pjsip: Add rtptimeout support
ASTERISK-25260: Wiki Documentation - Parking!
ASTERISK-25261: Manager events for MeetMe have incorrectly documented key name 'Usernum' - should be 'User'
ASTERISK-25262: Memory leak when a caller channel does multiple dials and CEL is enabled
ASTERISK-25263: [patch]cdr_adaptive_odbc: CDR insert failure due to reversed if logic
ASTERISK-25264: Crash in ast_chan_setoption
ASTERISK-25265: [patch]DTLS Failure when calling WebRTC-peer on Firefox 39 - add ECDH support and fallback to prime256v1
ASTERISK-25266: Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
ASTERISK-25267: [patch] Alembic - broken compatibility for Oracle and Microsoft SQL
ASTERISK-25268: Neither a src change or marker after (attended) transfer
ASTERISK-25269: Speex VAD: SIGSEGV when softmixing stasis-bridges
ASTERISK-25270: chan_sip: rtptimeout doesn't work at all when using JitterBuffers of any kind
ASTERISK-25271: Parking & blind transfer: Transferer channel not hung up if no MOH
ASTERISK-25272: [patch]The ICONV dialplan function sometimes returns garbage
ASTERISK-25273: Can't compile Asterisk - 'make menuselect' fails with segfault and core dump
ASTERISK-25274: A11 SIGSEGV 'Double free or corruption' in backtrace from pj_pool_release (sip_destroy -> pj_ice_sess_destroy)
ASTERISK-25275: A11 SIGSEGV from pjnpath check_cached_response (ast_rtcp_read -> pj_stun_session_on_rx_pkt)
ASTERISK-25276: "confbridge record" makes # of asterisk subprocess growing
ASTERISK-25277: Asterisk forwards DTMF when call is on hold
ASTERISK-25278: pjsip: MWI aggregate test is failing consistently.
ASTERISK-25279: Deadlock using chan_sip
ASTERISK-25280: Reset service Asterisk: ERROR astobj2.c: user_data is NULL
ASTERISK-25281: PJSIP, ODBC and Oracle - case sensitive field name checks in sorcery break Oracle compatibility
ASTERISK-25282: rtptimeout does not kick in because of zero sized frames (lost packets)
ASTERISK-25283: res_ari: Multiple PJSIP contacts can't be dialed directly
ASTERISK-25284: Parking & blind transfer: Transferer channel not hung up if no MOH
ASTERISK-25285: app_voicemail prompts play improperly for Portuguese language
ASTERISK-25286: After a period of time chan_iax2 stops accepting packets
ASTERISK-25287: Busy Detect not working
ASTERISK-25288: Configure fails to detect uuid-devel on Fedora 22
ASTERISK-25289: Build System does not respect CFLAGS and CXXFLAGS when building menuselect
ASTERISK-25290: Build System does not respect CFLAGS and CXXFLAGS placed on the command line
ASTERISK-25291: Test suite not available via `make check` or `make test`
ASTERISK-25292: Testuite: tests/apps/bridge/bridge_wait/bridge_wait_e_options fails
ASTERISK-25293: Crash in DNS core if no DNS result set on query
ASTERISK-25294: srtp's crypto_get_random deprecated
ASTERISK-25295: res_pjsip crash - pjsip_uri_get_uri at /usr/include/pjsip/sip_uri.h
ASTERISK-25296: RTP performance issue with several channel drivers.
ASTERISK-25297: Crashes running channels/pjsip/resolver/srv/failover/in_dialog testsuite tests
ASTERISK-25298: pjsip: subscriptions/mwi/unsolicited/mailbox_count_changes sporadically failing
ASTERISK-25299: RTP port leaks with incoming OOH323 calls
ASTERISK-25300: ConfBridge() in 13.1-cert2 is not combining configuration from static text config file with dynamic config in dialplan with CONFBRIDGE function.
ASTERISK-25301: asterisk segfault in res_hep_pjsip.so on client connect
ASTERISK-25302: Error:Oh dear... we couldn't allocate a port for RTP instance
ASTERISK-25303: Error 4 in app_queue.so in 13.1-cert2
ASTERISK-25304: res_pjsip: XML sanitization may write past buffer
ASTERISK-25305: Dynamic logger channels can be added multiple times
ASTERISK-25306: Persistent subscriptions can save multiple SIP messages at once, leading to potential crashes.
ASTERISK-25307: Hangup on channel using FastAGI does not hang up child channels
ASTERISK-25308: ari: Websocket leak
ASTERISK-25309: [patch] iLBC 20 advertised
ASTERISK-25310: [patch]on FreeBSD also pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED
ASTERISK-25311: Asterisk may restart if using cdr_odbc + MySQL in strict mode
ASTERISK-25312: res_http_websocket: Terminate connection on fatal cases
ASTERISK-25313: tests/bridge/connected_line_update: Sporadically failing
ASTERISK-25314: tests/rest_api/asterisk/logging/get_logging: Consistently failing
ASTERISK-25315: DAHDI channels send shortened duration DTMF tones.
ASTERISK-25316: PJSIP qualify in mutlihomed system sent from wrong transport
ASTERISK-25317: asterisk sends too many stun requests
ASTERISK-25318: tests/rest_api/applications/subscribe-endpoint/nominal/resource: Sporadically failing
ASTERISK-25319: tests/rest_api/asterisk/logging/rotate_log / add_log: Sporadically failing
ASTERISK-25320: chan_sip.c: sip_report_security_event searches for wrong or non existent peer on invite
ASTERISK-25321: [patch]DeadLock ChanSpy with call over Local channel
ASTERISK-25322: Crash occurs when using MixMonitor with t() or r() options.
ASTERISK-25323: Asterisk: ongoing segfaults uncovered by CHAOS_DEBUG
ASTERISK-25324: MAKE CALL /Recieve Call from Web client in asterisk 1.8
ASTERISK-25325: ARI PUT reload chan_sip HTTP response 404
ASTERISK-25326: asterisk-ari: ARI API to stop running music
ASTERISK-25327: Annoying ERROR: lock.c:459 __ast_pthread_mutex_unlock: app_queue.c line 6445 (try_calling): mutex 'qe->chan' freed more times than we've locked!
ASTERISK-25328: Wrong ANSWEREDTIME after Dial app execution
ASTERISK-25329: Asterisk configure fails on 'cannot find ptlib-config', despite ptlib-config existing
ASTERISK-25330: Failed to originate a call using AMI on OOH323 channel (Avaya)
ASTERISK-25331: install_prereq is not installing sqlite 3 library on CentOS
ASTERISK-25332: marker bit lost in outgoing stream when incoming stream has vad
ASTERISK-25333: [patch]App meetme sometimes not close fd with pseudo module
ASTERISK-25334: Testing
ASTERISK-25335: Testing bob
ASTERISK-25336: Documentation: Explanation of DIALEDTIME and ANSWEREDTIME variables could be more explicit
ASTERISK-25337: Crash on PJSIP_HEADER Add P-Asserted-Identity when calling from Gosub
ASTERISK-25338: Failed to authenticate device messages don't report connection ip
ASTERISK-25339: res_pjsip: Empty "auth" sections from non-config backgrounds are interpreted as valid
ASTERISK-25340: Manager.conf TLS doesn't activates
ASTERISK-25341: bridge: Hangups may get lost when executing actions
ASTERISK-25342: res_pjsip: Repeated usage of pj_gethostip may block
ASTERISK-25343: Successful attended transfer on queue leaves agent with Music On Hold
ASTERISK-25344: Random crashes (SIGSEGV / SIGABRT) on Asterisk 13.1.0~dfsg-1+b1 (possibly in combination with stasis)
ASTERISK-25345: pjsip:0 <?>: tsx0xb3fe5d1c ...Failed to send Request msg INVITE/cseq=28532 (tdta0xb6bb49c0)! err=171060 (Unsupported transport (PJSIP_EUNSUPTRANSPORT))
ASTERISK-25346: chan_sip: Overwriting answered elsewhere hangup cause on call pickup
ASTERISK-25347: Crash on res_odbc reload in SOCK_flush_output from /usr/lib64/psqlodbc.so
ASTERISK-25348: Asterisk Crashes after caller records name with the Privacy Options
ASTERISK-25349: Asterisk freeze due to schedule ID cancellation failure
ASTERISK-25350: Asterisk crashes
ASTERISK-25351: SIP INFO not ACK'd by chan_pjsip on Asterisk 13.5.0
ASTERISK-25352: res_hep_rtcp correlation_id is different then res_hep
ASTERISK-25353: [patch] Transcoding while different in Frame size = Frames lost
ASTERISK-25354: Unable to compile any version
ASTERISK-25355: sched: ast_sched_del may return prematurely due to spurious wakeup
ASTERISK-25356: res_pjsip_sdp_rtp: Multiple keepalive scheduled items may exist
ASTERISK-25357: AMI GetConfigJSON
ASTERISK-25358: dateformat not read from logger.conf by remote console
ASTERISK-25359: Leak when not Answering
ASTERISK-25360: chan_sip:wrong ice candidate fails html5client on mobile connection
ASTERISK-25361: [patch]possible null pointer issue in chan_iax2 - with potential fix
ASTERISK-25362: Deadlock due to presence state callback
ASTERISK-25363: PJSIP/rls and Testsuite: channels/pjsip/subscriptions/rls/lists/off_nominal/large_notify reports success after reactor times out and terminates the unfinished sipp scenario
ASTERISK-25364: [patch]Issue a TCP connection(kernel) and thread of asterisk is not released
ASTERISK-25365: Persistent subscriptions have extra Content-Length/corrupted messages
ASTERISK-25366: Segmentation fault - in ast_manager_build_channel_state_string_prefix at manager_channels.c:417
ASTERISK-25367: pbx: Long pattern match hints may cause "core show hints" to crash
ASTERISK-25368: English sound prompt no-valid-responce-transfering has an incorrect name and is mis-matched vs other language sets
ASTERISK-25369: res_parking: ParkAndAnnounce - Inheritable variables aren't applied to the announcer channel
ASTERISK-25370: res_corosync segfaults at startup with corosync version > 2.x
ASTERISK-25371: Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts to generate hangup event
ASTERISK-25372: SIP/2.0 401 Unauthorized for incoming calls.
ASTERISK-25373: add documentation for CALLERID(pres) and also the CONNECTEDLINE and REDIRECTING variants
ASTERISK-25374: Crash in CDR handle_dial_message where peer is null
ASTERISK-25375: Bad ao2 pointer on snapshot cleanup after creation
ASTERISK-25376: Scripts: check file versions for Asterisk and dependencies
ASTERISK-25377: res_pjsip: Change default "From user" from UUID to something more palatable
ASTERISK-25378: Segfault in pj_pool_alloc () from /usr/lib/libpj.so.2
ASTERISK-25379: no sound on pjsip channel with bridge_native_rtp enabled
ASTERISK-25380: Oh dear... we couldn't allocate a port for RTP instance
ASTERISK-25381: res_pjsip: AoRs deleted via ARI (or other mechanism) do not destroy their related contacts
ASTERISK-25382: segfault in ast_json_free (p=0x77f9d88555b8) at json.c:190
ASTERISK-25383: Core dumps on startup and shutdown with MALLOC_DEBUG enabled
ASTERISK-25384: Regular Asterisk crashes when using Page application. "user_data is NULL"
ASTERISK-25385: A11 SIGSEGV Crashes in srtp_get_stream (), in ast_srtp_protect
ASTERISK-25386: Asterisk Chan_sip.c Deadlock. All SIP traffic stops
ASTERISK-25387: res_pjsip_nat: Malformed REGISTER request causes NAT'd Contact header to not be rewritten
ASTERISK-25388: chan_sip partial unresponsive, one permanent lock
ASTERISK-25389: pjsip: crash on null uri in contact header
ASTERISK-25390: default_from_user can crash with certain configuration backends
ASTERISK-25391: AMI GetConfigJSON returns invalid JSON
ASTERISK-25392: A11 Deadlock detected, full thread bt
ASTERISK-25393: Non-realtime and Realtime can not exist together
ASTERISK-25394: pbx: Incorrect device and presence state when changing hint details
ASTERISK-25395: Crash when establishing subscription with pjsip
ASTERISK-25396: chan_sip: Extremely long callerid name causes invalid SIP
ASTERISK-25397: [patch]chan_sip: File descriptor leak with non-default timert1
ASTERISK-25398: chan_iax2 call failed to authenticate
ASTERISK-25399: app_queue: AgentComplete event has wrong reason
ASTERISK-25400: Hints broken when "CustomPresence" doesn't exist in AstDB
ASTERISK-25401: Segmentation-fault crash within pjnath
ASTERISK-25403: A11 SIGSEGV clearerr (fp=0x0) at clearerr.c:26...in ast_careful_fwrite
ASTERISK-25404: segfault/crash in chan_pjsip_hangup ... at chan_pjsip.c
ASTERISK-25405: [patch] CLI: core show fd: add timestamp
ASTERISK-25406: Misc anomalies in Swagger definitions
ASTERISK-25407: Asterisk fails to log to multiple syslog destinations
ASTERISK-25408: One RTP stream is lost out of the NIC for approx 5 sec then returns
ASTERISK-25409: Asterisk not reading entire TLSCERTFILE
ASTERISK-25410: app_record: RECORDED_FILE variable not being populated
ASTERISK-25411: PJSIP functionality becomes unresponsive after some time
ASTERISK-25412: Blanks in database fields leads to unexpected results
ASTERISK-25413: res_pjsip: does not have IP only endpoint identification per-endpoint - indentify_by work-around
ASTERISK-25414: CLONE - [patch] IMS TEL URI incoming INVITE RFC 3966 not recognized
ASTERISK-25415: A11 SIGSEGV 'Double free or corruption' in backtrace from pj_pool_release
ASTERISK-25416: pbx_dundi: Using scheduler context after deletion
ASTERISK-25417: res_fax - Asterisk does not send localstationidentifier to remote site
ASTERISK-25418: On-hold channels redirected out of a bridge appear to still be on hold
ASTERISK-25419: Dialplan Application for Integration of StatsD
ASTERISK-25420: Some documentation issues for AMI
ASTERISK-25421: PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending
ASTERISK-25422: [patch]The 'Bridge' event always reports the 'Bridgetype' as 'core' even if it's 'native'
ASTERISK-25423: Caller gets no Connected line update during call pickup.
ASTERISK-25424: asterisk.conf syntax error causes inscrutable crash
ASTERISK-25425: logger: Add JSON structured logging
ASTERISK-25426: Core dump in CDR handler
ASTERISK-25427: Callerid change does not always emit NewCallerid AMI event
ASTERISK-25428: Codec negotation fails 'No compatible codecs, not accepting this offer!' in transfer scenario between two servers when using SILK and SPEEX
ASTERISK-25429: res_pjsip_endpoint_identifier_ip: Document support for hostnames
ASTERISK-25430: Testsuite: batched rls subscription failure
ASTERISK-25431: A11 SIGSEGV in check_cached_response (ast_rtcp_read -> pj_stun_session_on_rx_pkt)
ASTERISK-25432: Testsuite: tests/channels/SIP/tcpauthlimit/tcp_client_scenario is Failing
ASTERISK-25433: Crash on CLI command 'data get'
ASTERISK-25434: Compiler flags not reported in 'core show settings' despite usage during compilation
ASTERISK-25435: Asterisk periodically hangs. UDP Recv-Q greatly exceeds zero.
ASTERISK-25436: Segmentation fault relating to JSON, stasis, and fax
ASTERISK-25437: segfault/crash in chan_pjsip_hangup ... at chan_pjsip.c
ASTERISK-25438: res_rtp_asterisk: ICE role message even when ICE is not enabled
ASTERISK-25439: Segfault in find_entry () from /usr/lib/libpj.so.2 (dns_resolver, qualify_contact)
ASTERISK-25440: Asterisk does not use owner parameter in SDP breaking RFC3264
ASTERISK-25441: Deadlock in res_sorcery_memory_cache.
ASTERISK-25442: using realtime (mysql) queue members are never updated in wait_our_turn function (app_queue.c)
ASTERISK-25443: [patch]IPv6 - Potential issue in via header parsing
ASTERISK-25444: [patch]Music On Hold Warning misleading
ASTERISK-25445: AMI QueueStatus broken
ASTERISK-25446: Warnings with jitter buffer enabled and transcoding from G722 to ulaw
ASTERISK-25447: Wrong template file 'realtime.sql' in contrib/realtime/postgresql
ASTERISK-25448: Motor Sound after pickup incoming call
ASTERISK-25449: main/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other potential scheduling issues in chan_sip/chan_skinny
ASTERISK-25450: Multiple realtime entries for same (ldap)driver in sorcery.conf not working anymore.
ASTERISK-25451: Broken video - erased rtp marker bit
ASTERISK-25452: Registering with Caret character at the end of password
ASTERISK-25453: user_agent not set in Server header
ASTERISK-25454: Crashing possibly related to cdr_manager
ASTERISK-25455: Deadlock of PJSIP realtime over res_config_pgsql
ASTERISK-25456: Asterisk becomes unresponsive - res_rtp_asterisk: WebRTC STUN Deadlock
ASTERISK-25457: Chan_PJSIP No MoH / Hold
ASTERISK-25458: Unable to set CDR variable in h extension or hangup_handler
ASTERISK-25459: New dependency on libxml2
ASTERISK-25460: UDP leak
ASTERISK-25461: Nested dialplan #includes don't work as expected.
ASTERISK-25462: pjsip show channels segfault: Address 0x2 out of bounds in res_sorcery_realtime.c
ASTERISK-25463: Starting Asterisk with ARI module started but sorcery not causes pain
ASTERISK-25464: Segfault with T.38 protocol and ReceiveFax Application
ASTERISK-25465: script for migrating realtime chan_sip tables to pjsip
ASTERISK-25466: pjsip: Endpoints added to pjsip.conf during runtime - reload results in an 'invalid' state for all but the last endpoint loaded
ASTERISK-25467: chan_sip/webrtc Asterisk + Chrome M47 consistent 0.9s ice handshake delay since commit 1ad827
ASTERISK-25468: Deadlock in chan_sip - core show locks shows do_monitor lock
ASTERISK-25469: chan_sip enabled unnegotiated session-timers after reINVITE if session-minse is non-default.
ASTERISK-25470: SDP version increased when no change in SDP on 183 session progress retransmit
ASTERISK-25471: [patch]Add subscribe_context to res_pjsip
ASTERISK-25472: Swagger scripts are not replacing format variable in file brief
ASTERISK-25473: Test infrastructure: Need to reimplement coverage reports
ASTERISK-25474: Test infrastructure: Need to reimplement REF_DEBUG testing
ASTERISK-25475: PJSIP tab completion of 'pjsip show endpoint' results in query storm and takes a very long time
ASTERISK-25476: chan_sip loses registrations after a while
ASTERISK-25477: pjsip show "command" like [criteria]
ASTERISK-25478: Need configuration option(s) to control how res_pjsip_endpoint_identifier_user performs endpoint lookup
ASTERISK-25479: Allow CDR's to be modified before being dispatched to engines
ASTERISK-25480: [patch]Add field PauseReason on QueueMemberStatus
ASTERISK-25481: res_pjsip listens on undefined UDP port, even with no transports configured
ASTERISK-25482: Faxes are randomly not sent using T.38 when faxing over a local channel
ASTERISK-25483: [patch] Built-in sounds are send at 40,20,40,20ms intervals for iLBC
ASTERISK-25484: [patch] autoframing=yes has no effect
ASTERISK-25485: res_pjsip_outbound_registration: registration stops due to 400 response
ASTERISK-25486: res_pjsip: Fix deadlock when validating URIs
ASTERISK-25487: chan_ooh323: Error Decoding H245 Message
ASTERISK-25488: Mute function issue with Asterisk ARI: Channel not in stasis.
ASTERISK-25489: Crash while calling function iax2_frame_free
ASTERISK-25490: [patch]SDP crypto tag is validated incorrectly
ASTERISK-25491: extenpatternmatchnew=yes breaks with hint and pattern match extension in the same context
ASTERISK-25492: ARI: Path parameters are case sensitive
ASTERISK-25493: voicemail email limited to 80 characters
ASTERISK-25494: build: GCC 5.1.x catches some new const, array bounds and missing paren issues
ASTERISK-25495: [patch] Prevent old-update packages on repository Debian systems
ASTERISK-25496: Random lockups using ARI
ASTERISK-25497: Calling fflush on stdout blocking
ASTERISK-25498: Asterisk crashes when negotiating g729 without that module installed
ASTERISK-25499: G722 Codec and Inband DTMF Support
ASTERISK-25500: voicemail update problem in PG when change password
ASTERISK-25501: Dial processing fail if there is comment like ;--==
ASTERISK-25502: res_pjsip_pubsub: Subscription not terminated on NOTIFY error responses
ASTERISK-25503: Compilation failure in Fedora 23 - 'collect2: error: ld returned 1 exit status'
ASTERISK-25504: Asterisk with pjsip driver crashes codec related?
ASTERISK-25505: res_pjsip_pubsub: Crash on off-nominal when UAS dialog can't be created
ASTERISK-25506: [patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
ASTERISK-25507: Dual channel redirect causes Volume audiohook to fail to work
ASTERISK-25508: Defaults cdr.conf.sample
ASTERISK-25509: Testsuite: tests/rest_api/channels/redirect/nominal Crash
ASTERISK-25510: [patch]Log to syslog failing
ASTERISK-25511: Missing Documentation for AMI Events QueueParams and QueueMember
ASTERISK-25512: Crash every day three times on vmWare
ASTERISK-25513: Crash: malloc failed with high load of subscriptions.
ASTERISK-25514: Can'nt compile
ASTERISK-25515: Testsuite: Handle when AMIFactory does not support reconnections
ASTERISK-25516: Too many objects of the specified type (PJ_ETOOMANY). Failed to send Request msg OPTIONS.
ASTERISK-25517: PCMA codec negotiated.. Asterisk unexpectedly uses G722 for first second or so of call before sending PCMA
ASTERISK-25518: taskprocessor: Add high water mark
ASTERISK-25519: Confbridge muted user gets unmuted while audio is playback on same conference room
ASTERISK-25520: ARI DELETE /bridges/{bridgeId} when recording doesn't delete
ASTERISK-25521: Official web page to read general information about asterisk in spanish?
ASTERISK-25522: ARI: Crash when creating channel via ARI originate with requesting channel
ASTERISK-25523: res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
ASTERISK-25524: module reload res_calendar.so does not reload everything in calendar.conf
ASTERISK-25525: A call EndWhile can cause the dial plan to jump to another context.
ASTERISK-25526: In case of incorrect request in a database, after several attempts, service the asterisk is disconnected
ASTERISK-25527: Quirky xmldoc description wrapping
ASTERISK-25528: DNS: System resolver issues with TTL parse
ASTERISK-25529: (Adaptive)CDR with MySQL Storing LinkedID in Uniqueid column
ASTERISK-25530: Asterisk 13.6 not presenting ami hangup event for queue calls
ASTERISK-25531: [patch] add debug detailing the location of a file search for Playback or Background
ASTERISK-25532: Asterisk crash on certain extension
ASTERISK-25533: [patch] buffer for ast_format_cap_get_names only 64 bytes
ASTERISK-25534: Core dump on RTCP report/DNS lookup
ASTERISK-25535: [patch] format creation on module load instead of cache
ASTERISK-25536: Testsuite: Sporadic failures in Attended transfer tests using 3PCC
ASTERISK-25537: [patch] format-attribute module: RFC or internal defaults?
ASTERISK-25538: [patch]Missing PID in syslog logger messages
ASTERISK-25539: 488 Not acceptable here sent after T.38 re-invite accepted
ASTERISK-25540: Core dump related to RTCP report
ASTERISK-25541: add socket activation support
ASTERISK-25542: log to journald directly
ASTERISK-25543: replace openssl with GnuTLS
ASTERISK-25544: remove upstart from contrib
ASTERISK-25545: [patch] translation module gets cached not joint format
ASTERISK-25546: threadpool: Race condition between idle timeout and activation
ASTERISK-25547: Asterisk crashed when user came in confbridge
ASTERISK-25548: stasis: Improve message type "Use of before init/after destruction" error
ASTERISK-25549: Confbridge: Add participant timeout option
ASTERISK-25550: Codecs negotiated incorrectly
ASTERISK-25551: [patch]Ability to add channel to an existing bridge by specifying an existing channel prefix
ASTERISK-25552: hashtab: Improve NULL tolerance
ASTERISK-25553: endpoints created via pjsip_wizard does not have registrations
ASTERISK-25554: confusing module loading errors on startup
ASTERISK-25555: Failing Testsuite Test: 'rls/lists/nominal/presence/full_state'
ASTERISK-25556: Testsuite: Test Failure Detected for 'rls/lists/nominal/presence/full_state'
ASTERISK-25557: Testsuite: Add Debug Message for Incorrect Module 'typename' Values Specified in 'test-config.yaml'
ASTERISK-25558: [patch]chan_sip option 'notifyringing' doc fix and addition of 'notifyringingprio'
ASTERISK-25559: Asterisk12 with webRTC , can ring but no audio and video
ASTERISK-25560: speex module fails to compile
ASTERISK-25561: app_queue.c line 6503 (try_calling): mutex 'qe->chan' freed more times than we've locked!
ASTERISK-25562: testsuite: Ability to per-test disable logging
ASTERISK-25563: Channel Hangup after DTMF during Playback
ASTERISK-25564: CHAOS: Assertion in ast_ari_callback
ASTERISK-25565: DNS: System resolver only returns 1 record per result
ASTERISK-25566: Double log entries
ASTERISK-25567: Log IP Addresses for automatic firewalling (e.g. fail2ban)
ASTERISK-25568: [patch]180 Ringing not sent after 183 Session Progress
ASTERISK-25569: app_meetme: Audio quality issues
ASTERISK-25570: DNS: Implement negative connection cache
ASTERISK-25571: PJSIP: Add StatsD stats for some common PJSIP objects
ASTERISK-25572: Endpoints: Add StatsD stats for Asterisk endpoints
ASTERISK-25573: [patch] H.264 format attribute module: resets whole SDP
ASTERISK-25574: OPening TCP Port 5060
ASTERISK-25575: res_pjsip: Dynamic outbound registrations created via ARI are not loaded into memory on Asterisk start/restart
ASTERISK-25576: Crash while cancelling thread in chan_skinny
ASTERISK-25577: Crash while walking thread list to unregister a thread on shutdown
ASTERISK-25578: [patch] SIP/SDP: No rtpmap for static RTP payload IDs
ASTERISK-25579: Outgoing Packetization Time (Speex, AMR, Opus, …)
ASTERISK-25580: WSS will not work with asterisk 11
ASTERISK-25581: [patch]Add value reason a pause on CLI
ASTERISK-25582: Testsuite: Reactor timeout error in tests/fax/pjsip/directmedia_reinvite_t38
ASTERISK-25583: [patch] format-attribute module: RFC 7587 (Opus Codec)
ASTERISK-25584: [patch] format-attribute module: VP8 missing
ASTERISK-25585: [patch]rasterisk never hits most of main(), but it's assumed to
ASTERISK-25586: uuid_generate_random detection failure
ASTERISK-25587: Libedit2 colored prompt is broken beyond repair
ASTERISK-25588: Problem exchanging device states with PJSIP
ASTERISK-25589: Improve documentation on transport configuration
ASTERISK-25590: CLI Usage info for 'pjsip send notify' references incorrect config
ASTERISK-25591: [patch] Complete List of Header Files (#include): iwyu
ASTERISK-25592: chan_unistim: Clang Warning: variable sized type not at end of a struct
ASTERISK-25593: fastagi: record file closed after sending result
ASTERISK-25594: Queue strategy linear
ASTERISK-25595: Unescaped : in messge sent to statsd
ASTERISK-25596: CDR engine dispatching 2 cdrs - one for PartyA and another combined PartyA-PartyB
ASTERISK-25597: Remote console freeze after 'core stop gracefully' and then further command attempts
ASTERISK-25598: res_pjsip: Contact status messages are printing a hash instead of the uri
ASTERISK-25599: [patch] SLIN Resampling Codec only 80 msec
ASTERISK-25600: bridging: Inconsistency in BRIDGEPEER
ASTERISK-25601: json: Audit reference usage and thread safety
ASTERISK-25602: chan_sip deadlocks after INVITE processing while calling sip_report_security_event
ASTERISK-25603: [patch]udptl: Uninitialized lengths and bufs in udptl_rx_packet cause ast_frdup crash
ASTERISK-25604: Dial application 'r' argument does not initiate ringing on a Motif channel
ASTERISK-25605: outboundproxy= works globally but not for individual SIP trunks
ASTERISK-25606: Core dump when using transports in sorcery
ASTERISK-25608: res_pjsip/contacts/statsd: Lifecycle events aren't consistent
ASTERISK-25609: [patch]Asterisk may crash when calling ast_channel_get_t38_state(c)
ASTERISK-25610: Asterisk crash during "sip reload"
ASTERISK-25611: core: threadpool thread_timeout_thrash unit test sporadically failing
ASTERISK-25612: Configuration parser handles unsigned integers as signed integers
ASTERISK-25613: Documentation: ConnectedLineNum / Name same value as CallerID if call started from ORIGINATE
ASTERISK-25614: DTLS negotiation delays
ASTERISK-25615: res_pjsip: Setting transport async_operations > 1 causes segfault on tls transports
ASTERISK-25616: Warning with a Codec Module which supports PLC with FEC
ASTERISK-25617: Asterisk 11 segfaults in pj_stun_session_on_rx_pkt
ASTERISK-25618: res_pjsip: Check for readability of TLS files at startup
ASTERISK-25619: res_chan_stats not sending the correct information to StatsD
ASTERISK-25620: Call pickup during a Multi-party Dial results in a channel hanging up later than it should and a duplicate CDR entry.
ASTERISK-25621: res_pjsip: outbound_proxy arbitrarily and occasionally set to 'asterisk' during reload
ASTERISK-25622: WARNING for "JABBER: socket read error" should be more specific
ASTERISK-25623: "><img src=1 onerror=alert(1)>
ASTERISK-25624: AMI Event OriginateResponse bug
ASTERISK-25625: res_sorcery_memory_cache: Add full backend caching
ASTERISK-25626: Integrating Outbound Routes with context
ASTERISK-25627: Easily Preventable Compile Warning
ASTERISK-25628: res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
ASTERISK-25629: [patch] Native Packet-Loss Concealment (PLC)
ASTERISK-25630: ari: Creating bridge with id of one that exists outside of ARI doesn't cause error
ASTERISK-25631: segfault at rtp_engine.c:1525
ASTERISK-25632: res_pjsip_sdp_rtp: RTP is sent from wrong IP address when multihomed
ASTERISK-25633: chan_sip: Codec preference misorder
ASTERISK-25634: How are updates installed in AsteriskNOW
ASTERISK-25635: run_agi() while() loop loops indefinitely because of fgets() returns EAGAIN
ASTERISK-25636: delete user lakatmester
ASTERISK-25637: Multi homed server using wrong IP
ASTERISK-25638: pjsip: Deadlock between monitor thread and worker threads
ASTERISK-25639: app_amd: system maxwords discrepency
ASTERISK-25640: pbx: Deadlock on features reload and state change hint.
ASTERISK-25641: bridge: GOTO_ON_BLINDXFR doesn't work on transfer initiated channel
ASTERISK-25642: res_rtp_asterisk: SRTCP broken with DTLS - bad video is one of the consequences
ASTERISK-25643: Internal Chan / PJSIP Cause code related Crash
ASTERISK-25644: No event in queue log when member leaves the queue
ASTERISK-25645: res_rtp_asterisk: Lock inversion
ASTERISK-25646: CLI output after "core stop gracefully" on a remote console is confusing and inconsistent with root console behavior
ASTERISK-25647: bug of cel_radius.c: wrong point of ADD_VENDOR_CODE
ASTERISK-25648: chan_sip returns forbidden 403, if the incoming number was determined as the present.
ASTERISK-25649: Transfer application does not work with Local channels - documentation misleading
ASTERISK-25650: Crash before unload res_pjsip_exten_state.so and load again res_pjsip_exten_state.so
ASTERISK-25651: Error loading several modules in Fedora 23 with 'undefined symbol' - is modules.conf missing dependencies?
ASTERISK-25652: func_curl: Add the ability to CURL files down to a specified location
ASTERISK-25653: Deadlock - PJ_ENOMEM errors & high Recv-Q counts when using PJSIP TLS extensions
ASTERISK-25654: Playback: Add the ability to play remote URIs
ASTERISK-25655: core taskprocessors sorcery-control memory not cleared
ASTERISK-25656: 183 is missing codec mappings with Chan SIP and Asterisk 13 when transcoding required - REINVITE occuring when it shouldn't, also missing codec mappings.
ASTERISK-25657: pbx: Split up logical parts of the PBX core into separate things
ASTERISK-25658: Random segmentation fault for asterisk webrtc
ASTERISK-25659: res_rtp_asterisk: ECDH not negotiated causing DTLS failure occurred on RTP instance
ASTERISK-25660: Add sipp-sendfax.xml and spandspflow2pcap.py to contrib/scripts.
ASTERISK-25661: No clean way to stop Playback in Asterisk 13 and PLAYBACKSTATUS regression
ASTERISK-25662: Malformed AGI 520 Usage response
ASTERISK-25663: app_queue: Segfault in queue during playback of periodic announcement - filename address out of bounds
ASTERISK-25664: ast_format_cap_append_by_type leaks a reference
ASTERISK-25665: Duplicate logging in queue log for EXITEMPTY events
ASTERISK-25666: chan_sip: Path header is ignored
ASTERISK-25667: Subscription request from endpoint XXX rejected. Expiration of 0 is invalid
ASTERISK-25668: res_pjsip: Deadlock in distributor
ASTERISK-25669: [patch]CURL incorrect trim for non ASCII characters
ASTERISK-25670: Add regcontext to PJSIP
ASTERISK-25671: Asterisk often gets a SIGSEGV, Segmentation fault
ASTERISK-25672: 13.6 PJSIP Crash on Startup
ASTERISK-25673: res_crypto leaks CLI entries
ASTERISK-25674: Mixmonitor stop recording after atxfer
ASTERISK-25675: Endpoint not listed as Unreachable
ASTERISK-25676: chan_sip: Codecs on RTP instance are all offered, not combined
ASTERISK-25677: pbx_dundi: leaks during failed load.
ASTERISK-25678: app_confbridge: Add list concise command
ASTERISK-25679: res_calendar leaks scheduler.
ASTERISK-25680: manager: manager_channelvars is not cleaned at shutdown
ASTERISK-25681: devicestate: Engine thread is not shut down
ASTERISK-25682: Unable to Make with imap support
ASTERISK-25683: res_ari: Asterisk fails to start if compiled with MALLOC_DEBUG
ASTERISK-25684: codec: Translation of slin16 results in noise
ASTERISK-25685: infrastructure: Run alembic in Jenkins build script
ASTERISK-25686: PJSIP: qualify_timeout is a double, database schema is an integer
ASTERISK-25687: res_musiconhold: Concurrent invocations of 'moh reload' cause a crash
ASTERISK-25688: configure: No check for PJSIP pj_timer_entry_running
ASTERISK-25689: pjsip show contacts not working in Asterisk 13.7rc2
ASTERISK-25690: Hanging up when executing connected line sub does not cause hangup
ASTERISK-25691: Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
ASTERISK-25692: app_queue: shared_lastcall does not work when using realtime queues and Local channels
ASTERISK-25693: cdr_pgsql: Refactoring
ASTERISK-25694: cel_pgsql: Refactoring
ASTERISK-25695: safe_asterisk -c breaks color in asterisk -r because of missing TERM for /dev/tty.
ASTERISK-25696: bridge_basic: don't cache xferfailsound during a transfer
ASTERISK-25697: bridge_basic: don't play an attended transfer fail sound after target hangs up
ASTERISK-25698: Asterisk stopped
ASTERISK-25699: Segfault in check_cached_response
ASTERISK-25700: main/config: Clean config maps on shutdown.
ASTERISK-25701: core: Endless loop in "core show taskprocessors"
ASTERISK-25702: PjSip realtime DB and Cache Errors since upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2
ASTERISK-25703: chan_sip: externrefresh (sip.conf) isn't working
ASTERISK-25704: pjsip show settings shows debug=no no matter value in debug setting in pjsip.conf
ASTERISK-25705: PJSIP debug settings are inconsistent between console and file output
ASTERISK-25706: pbx: Abort asterisk on features reload (handle_hint_change)
ASTERISK-25707: Long contact URIs or hostnames can crash pjproject/Asterisk under certain conditions
ASTERISK-25708: chan_iax2: No ActionID field in iaxpeers Asterisk Manager command
ASTERISK-25709: ARI: Crash can occur due to race condition when attempting to operate on a hung up channel
ASTERISK-25710: core: Crash when publishing message that variable has been set
ASTERISK-25711: after answering tranfering the call iam disconecting the call ..but still my status is showing as queue
ASTERISK-25712: Second call to already-on-call phone and Asterisk sends "Ready"
ASTERISK-25713: Queue produces large number of undesired cdr records
ASTERISK-25714: ASAN:heap-buffer-overflow in logger.c
ASTERISK-25715: [patch] ASAN:global-buffer-overflow pjsip
ASTERISK-25716: Documentation: Document explanations and examples for possible values of DIALSTATUS
ASTERISK-25717: ASAN in most installed libsrtp
ASTERISK-25718: file: Use after free during shutdown
ASTERISK-25719: Direct media failure and strange logger output - similar failures with chan_sip or res_pjsip
ASTERISK-25720: core: Make channel variable allocation easier to read
ASTERISK-25721: [patch] res_phoneprov: memory leak and heap-use-after-free
ASTERISK-25722: ASAN & testsute: stack-buffer-overflow in sip_sipredirect
ASTERISK-25723: crash on dial with option p or P (privacy mode)
ASTERISK-25724: Many memory leaks and few asan bugs
ASTERISK-25725: core: Incorrect XML documentation may result in weird behavior
ASTERISK-25726: Asterisk compilation fails: 'redhat-hardened-ld: no such file or directory' - Asterisk on Fedora 23 appears to require redhat-rpm-config
ASTERISK-25727: RPM build requires OPTIONAL_API cflag due to PJSIP requirement
ASTERISK-25728: Better tracking of calls within Asterisk Logs needed (UNIQUEID)
ASTERISK-25729: [patch] Extension to device state translations are missing some extension states
ASTERISK-25730: build: make uninstall after make distclean tries to remove root
ASTERISK-25731: When a queue agent transfers a queue call, wrapuptime is not respected
ASTERISK-25732: [patch] persist queue member pause reason through restart
ASTERISK-25733: Called with SDP without ice-ufrag and ice-pwd
ASTERISK-25734: Mixmonitor - gaps in recordings
ASTERISK-25735: [patch] res_xmpp: Does not connect in component mode
ASTERISK-25736: pbx core: Deadlock during a reload
ASTERISK-25737: res_pjsip_outbound_registration: line option not in Alembic
ASTERISK-25738: res_pjsip_pubsub: Crash while executing OutboundSubscriptionDetail ami action
ASTERISK-25739: IAX2 - No native bridge on peers when encryption=yes
ASTERISK-25740: Asterisk aborts after receiving a MySQL syntax error for adaptive_odbc CDR logging
ASTERISK-25741: res_pjsip: "Contact" contains UUID for user portion
ASTERISK-25742: Secondary IFP Packets can result in accessing uninitialized pointers and a crash
ASTERISK-25743: Registered Peers Goes Unregistered
ASTERISK-25744: res_pjsip: Segfaults in ssl3_write_bytes, pj_ssl_sock_send, tls_send_msg
ASTERISK-25745: Crash in append_history_va
ASTERISK-25746: func_odbc: Requires an additional ARG when executing via CLI and Dialplan
ASTERISK-25747: Crash on "restart when convenient"
ASTERISK-25748: No audio h323 Asterisk with Ericsson MD110
ASTERISK-25749: StatsD dialplan application not existes
ASTERISK-25750: features: Crash occurs when executing a "features reload"
ASTERISK-25751: res_pjsip: Support pjsip_dlg_create_uas_and_inc_lock
ASTERISK-25752: Register 2 ext cisco 9951 into 2 separate asterisk server
ASTERISK-25753: No remote Party ID in PJSIP header invite
ASTERISK-25754: install_prereq doesn't work in Ubuntu 15.10
ASTERISK-25756: INVITEs are sent to old peer ip address if usereqphone is set true
ASTERISK-25757: When the Agents press hold in Softphone (X-litle) automatically the external call are hangup and the System put the Agents logoff.
ASTERISK-25758: The Callerid don´t appears on X-litle and many Agents after finished is time work cannot logoff
ASTERISK-25760: SayUnixTime aborts when it tries to give the time in French
ASTERISK-25761: USAN: Potential runtime errors causing undefined behavior
ASTERISK-25762: TSAN: Data race json unref
ASTERISK-25763: TSAN: Data race in json free
ASTERISK-25764: TSAN: low potencial data race in sig_flags
ASTERISK-25765: TSAN: data races and lock-order-inversions (potential deadlocks)
ASTERISK-25766: [patch] USAN can be used together with other sanitizers
ASTERISK-25767: [patch] Add check to configure for sanitizes
ASTERISK-25768: astobj2.c:124 INTERNAL_OBJ: bad magic number [..] for object [..]
ASTERISK-25769: Asterisk 13 is unable to convert audio files to sln48 format
ASTERISK-25770: Check for OpenSSL defines before trying to use them.
ASTERISK-25771: ARI:Crash - Attended transfers of channels into Stasis application.
ASTERISK-25772: res_pjsip: Unexpected two BYE when answered
ASTERISK-25773: ast flags macros broke atomic thread safe
ASTERISK-25774: Data race on deleting threads
ASTERISK-25775: stasis: Race condition with lock destruction in JSON usage
ASTERISK-25776: lock-order-inversion (potential deadlock) when loading app_queue
ASTERISK-25777: data race in threadpool
ASTERISK-25778: lock-order-inversion (potential deadlock) in res_pjsip
ASTERISK-25779: pjproject: Data race in pj_time
ASTERISK-25780: stasis: Potential deadlock
ASTERISK-25781: res_stun_monitor: Potential data race when accessing data
ASTERISK-25782: data race on logger
ASTERISK-25783: data race in ast_begin_shutdown
ASTERISK-25784: lock-order-inversion (potential deadlock) on dialplan reload
ASTERISK-25785: PJSIP Segmentation fault.
ASTERISK-25786: How to program with FreeVoip an incoming call on FXO rigning a FXS device with callerID info
ASTERISK-25787: chan_sip: Race condition when executing "sip show peers"
ASTERISK-25788: chan_sip: Race condition when executing "sip reload"
ASTERISK-25789: cdr_adaptive_odbc: Not storing custom fields in all cases
ASTERISK-25790: Unable to set X-P-Asserted-Identity with PJSIP_HEADER - the header doesn't make it into the SIP packet.
ASTERISK-25791: res_pjsip_caller_id: Lack of support for Anonymous <anonymous@anonymous.invalid>
ASTERISK-25792: chan_sip: qualifygap bounds checking
ASTERISK-25793: chan_sip: Incoming SIP packets ignored
ASTERISK-25794: Chan_sip allocates RTP ports even for rejected calls
ASTERISK-25795: Inbound fax is not working for L3 numbers when sent through Ring central account
ASTERISK-25796: res_pjsip: DOS/Crash when TCP/TLS sockets exceed pjproject PJ_IOQUEUE_MAX_HANDLES
ASTERISK-25797: app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
ASTERISK-25798: Agent CLI commands no longer exist on 11.21.2
ASTERISK-25799: can't place calls to throught sip trunk to cisco
ASTERISK-25800: [patch] Calculate talktime when is first call answered
ASTERISK-25801: app_mixmonitor: Does not continue after attended transfer
ASTERISK-25802: Segfault in rtp_engine.c
ASTERISK-25803: [patch] chan_sip: Optionally supply fromuser/fromdomain in SIP dial string
ASTERISK-25804: asterisk crash related to pjsip endpoint reginstration
ASTERISK-25805: Wiki documentation: Add a basic DUNDi how-to
ASTERISK-25806: Deep ACD queueing with app_queue.c causes denial of service on systems with a high core count by excessive thread mutex locks
ASTERISK-25807: Asterisk & WebRTC with DTLS-SRTP
ASTERISK-25808: Failed unpause update of realtime queue member
ASTERISK-25809: testsuite: tests/bridge/atxfer_fail_blonde fails or can give a false pass
ASTERISK-25810: say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
ASTERISK-25811: Unable to delete object from sorcery cache
ASTERISK-25812: res_pjsip_t38: Channel cannot do direct media after T.38
ASTERISK-25813: res_config_mysql dbsock parameter
ASTERISK-25814: Segfault at f ip in res_pjsip_refer.so
ASTERISK-25815: PJSIP RFC3323
ASTERISK-25816: French conf-adminmenu, conf-usermenu prompts differ in content from the English files
ASTERISK-25817: chan_sip: Keep alive messages contain trailing null byte
ASTERISK-25818: Blank extensions.conf, refuses to perform "dialplan save"
ASTERISK-25819: AMI hangup "cause" value ignored or overridden when channel is hungup during process of origination
ASTERISK-25820: Segmentation fault in ast_channel_dialed_causes_add
ASTERISK-25821: Segmentation fault in raise()
ASTERISK-25822: Segmentation fault in _int_malloc()
ASTERISK-25823: SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
ASTERISK-25824: How release a call if Receive a specific reason (e.g. Q.850;cause=7) in 183 Session Progress message
ASTERISK-25825: Crashes during shutdown when running CLI commands
ASTERISK-25826: PJSIP / Sorcery slow load from realtime
ASTERISK-25827: crash asterisk with dialplan add extension
ASTERISK-25828: Compile failure with older pjproject versions
ASTERISK-25829: res_pjsip: PJSIP does not accept spaces when separating multiple AORs
ASTERISK-25830: Revision 2451d4e breaks NAT
ASTERISK-25831: [patch] CEL MongoDB Backend
ASTERISK-25832: chan_sip: "Unknown peer" registration error does not retry
ASTERISK-25833: Asterisk 13.7.2 crashes with error about libmysqlclient.so.18.0.0
ASTERISK-25834: [patch] cdr_adaptive_odbc not logger when load before than database
ASTERISK-25835: Authentication using 'Username' field from Digest
ASTERISK-25836: Realtime MoH not working
ASTERISK-25837: file: Blocking when using FIFO
ASTERISK-25838: elastrix
ASTERISK-25839: "Expected to acknowledge ticks" problem
ASTERISK-25840: Asterisk 13.7.0 unable to send INVITEs to jsSIP (WebRTC) peer connected over WSS
ASTERISK-25841: pbx_spool fails
ASTERISK-25842: Move from linked lists to another type
ASTERISK-25843: chan_sip: Registration passwords can not contain @
ASTERISK-25844: app_queue: Ghost channels in "core show channels" output
ASTERISK-25845: res_pjsip_sdp_rtp: Wrong audio codec used when video enabled
ASTERISK-25846: Gracefully deal with Absent Stasis Apps
ASTERISK-25847: SLA causing segfaults
ASTERISK-25848: app_queue: Wrong channel in CONNECT and COMPLETECALLER events when call pickup feature code is used
ASTERISK-25849: chan_pjsip: transfers with direct media sometimes drops audio
ASTERISK-25850: Channel related AMI messages are arriving after the CHAN_END and LINKEDID_END CEL messages.
ASTERISK-25851: Bug in chan_sip - Forbidden 403
ASTERISK-25852: chan_iax2: Exceptionally long voice queue length with trunking
ASTERISK-25853: segfault in libpjnath.so.2
ASTERISK-25854: No audio after HOLD/RESUME - incorrect a=recvonly in SDP from Asterisk
ASTERISK-25855: No application AgentLogin after a while of operation
ASTERISK-25856: chan_sip: Path header is ignored (re-opening)
ASTERISK-25857: func_aes: incorrect use of strlen() leads to data corruption
ASTERISK-25858: [patch]Early media not processed when received ACM with some conditions
ASTERISK-25859: Asterisk not processing the calls
ASTERISK-25860: app_mixmonitor: Sound distortion on Playback
ASTERISK-25861: Subscription timeout earlier than anticipated - Asterisk accepts SUBSCRIBE with Event: presence and Accept: application/dialog-info+xml
ASTERISK-25862: No support for dynamic payload types in direct media
ASTERISK-25863: Occasionally Asterisk crashes when a iax2 channels connects
ASTERISK-25864: chan_sip: Error when trying to handle reINVITE from Asterisk in Chrome (DTLS-SRTP related)
ASTERISK-25865: Message-Account Missing From PJSIP MWI
ASTERISK-25866: ChanSpy: allow usage of a long queue to store audio frames, to avoid audio loss
ASTERISK-25867: [patch] Video delay on app_echo
ASTERISK-25868: Sorcery "append to category" should allow filters
ASTERISK-25869: chan_sip: "rejected because extension not found" should be logged as a security event
ASTERISK-25870: Deadlock while using Asterisk over mobile networks
ASTERISK-25871: Asterisk deadlock when using confbridge
ASTERISK-25872: logging to syslog despite my configuring not to
ASTERISK-25873: res_pjsip: Bundled pjproject: compile error, cannot find -lasteriskpj
ASTERISK-25874: app_voicemail: Stack buffer overflow in test_voicemail_notify_endl
ASTERISK-25875: Testsute can't detect builded version of sipp
ASTERISK-25876: Wiki Documentation - Configuration/Dialplan/Expressions
ASTERISK-25877: Wiki Documentation - Configuration/Applications/Bridge Application - Examples!
ASTERISK-25878: Wiki Documentation - Configuration/Applications/FollowMe - create, provide examples
ASTERISK-25879: Wiki Documentation - Configuration/Applications/Voicemail - Update, add more detail on config options for voicemail.conf, restructure and cleanup IMAP section
ASTERISK-25880: Wiki Documentation - Configuration/Applications/Queue - Rewrite/add tutorials and examples, remove AEL guide?
ASTERISK-25881: pbx: Add support for autohints
ASTERISK-25882: ARI: Crash can occur due to race condition when attempting to operate on a hung up channel (Part 2)
ASTERISK-25883: In the func_odbc doesn't work the option "escapecommas"
ASTERISK-25884: unable to ./configure after ./bootstrap.sh
ASTERISK-25885: res_pjsip: Race condition between adding contact and automatic expiration
ASTERISK-25886: Not checking for NULL INTERNAL_OBJ object, Asterisk crashes
ASTERISK-25887: ARI: POST "/channels/{channelId}/continue" inside Gosub - starts at priority n-1, repeats Stasis call before "continuing"
ASTERISK-25888: Frequent segfaults in function can_ring_entry() of app_queue.c
ASTERISK-25889: ARI: Add separate "create" and "dial" operations for channels
ASTERISK-25890: Asterisk 13.8.0 alembic database update fails
ASTERISK-25891: res_odbc: Crash using ODBC in mysql with heavy usage
ASTERISK-25892: I am planning to use VMware or KVM for installation of Asterisk using PRI card, kindly let me know does Asterisk support vmware environment
ASTERISK-25893: Function vmauthenticate accesses uninitialized memory
ASTERISK-25894: [patch] webrtc video broken due to missing marker bits in RTP streams
ASTERISK-25895: ODBC configuration not working with Postgres on Debian 8.4_64
ASTERISK-25896: app_voicemail: Option 'pollmailboxes' no longer working
ASTERISK-25897: RTCP feedback broken for video streams
ASTERISK-25898: Wiki Documentation: Configuration/Functions - create overview and a few child pages discussing the most commonly used functions
ASTERISK-25899: IMAP access FATAL error: Out of memory
ASTERISK-25900: PJSIP Endpoint IP Access Controls
ASTERISK-25901: Add transport for outbound PUBLISH
ASTERISK-25902: res_sorcery_memory_cache: Crash on "sorcery memory cache expire" CLI command
ASTERISK-25903: PJSIP AMI Event ContactStatus: add Useragent and RegExpire
ASTERISK-25904: PJSIP: add contact.updated event
ASTERISK-25905: Memory leak during perf testing
ASTERISK-25906: Delete this please.
ASTERISK-25907: Wiki documentation: Configuration/Interfaces - fill in intro page with a thorough overview
ASTERISK-25908: Wiki documentation: Deployment - Guides for Asterisk configuration with NAT and firewalls - scope out
ASTERISK-25909: Wiki documentation: Deployment/Emergency Calling - Create content
ASTERISK-25910: pjproject: Via headers are not parsed when "received" contains an IPv6 address
ASTERISK-25911: chan_iax2: IAX Max Retries - hung IAX channels in Ring state - cannot clear channels until Asterisk restart
ASTERISK-25912: chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding them to the local hangupcauses via ast_channel_hangupcause_hash_set
ASTERISK-25913: firends and peers cannot connect
ASTERISK-25914: PJSIP: failed registration with wrong codec name on allow/disallow
ASTERISK-25915: asterisk error can't have an outgoing call
ASTERISK-25916: Configuration file processing aborts if a configuration include target does not exist
ASTERISK-25917: [patch]app_voicemail: passwordlocation=spooldir only works if you manually add secret.conf yourself
ASTERISK-25918: SIP INFO (Key frame requests) not forwarded on
ASTERISK-25919: Duplicate DTMF events in ARI
ASTERISK-25920: Asterisk 13.8.0 segfaults using app.queue when ringinuse set to yes and another call comes in.
ASTERISK-25921: res_odbc: Crash of system in case of function invocation of ODBC
ASTERISK-25922: res_pjsip_exten_state: Add configuration support for publishing
ASTERISK-25923: Problem with download
ASTERISK-25924: chan_pjsip: Polycom SRTP problem
ASTERISK-25925: Allow Early Bridges on ARI Dials
ASTERISK-25926: DAHDI PRI calls echo only when no caller ID is present
ASTERISK-25927: Removed option "registertrying" is still documented in sip.conf.sample
ASTERISK-25928: res_pjsip: URI validation done outside of PJSIP thread
ASTERISK-25929: res_pjsip_registrar: AOR_CONTACT_ADDED events not raised
ASTERISK-25930: PJSIP: disable multi domain to improve realtime performace
ASTERISK-25931: PJSIP: add "reg_server" to contacts.
ASTERISK-25932: Error in Ring Strategy "leastrecent" using Dynamic Agents
ASTERISK-25933: res_pjsip_pubsub: Asterisk ignores expires header value
ASTERISK-25934: chan_sip should not require sipregs or updateable sippeers table unless rt
ASTERISK-25935: channels/sig_pri.h -> sig_pri_span -> force_restart_unavailable_chans in wrong scope
ASTERISK-25936: res_pjsip_dlg_options MODULEINFO section needs to be fixed
ASTERISK-25937: Send voicemail to yahoo accoun and local account both
ASTERISK-25938: res_odbc: MySQL/MariaDB statement LAST_INSERT_ID() always returns zero.
ASTERISK-25939: Program terminated with SEGV triggered by PJSIP_BYE_METHOD handler
ASTERISK-25940: AMI PlayDTMF plays DTMF in wrong thread
ASTERISK-25941: chan_pjsip: Crash on an immediate SIP final response
ASTERISK-25942: res_pjsip_caller_id: Transfer results in mixed ConnectedLine information
ASTERISK-25943: chan_local: Exceptionally long voice queue length queuing to Local/1323@internalexten-0001c0df;1
ASTERISK-25944: chan_sip: Multiple peers with same name
ASTERISK-25945: chan_pjsip: ABORT raised in channel_blob_dtor/ast_json_unref
ASTERISK-25946: queue_log being created even if app_queue is not loaded or existing
ASTERISK-25947: Protocol transfers to stasis applications are missing the StasisStart with the replace_channel object.
ASTERISK-25948: ast_pthread_mutex_lock calling ast_reentrancy_lock with lt=0x0
ASTERISK-25949: app_followme: FollowMe transmits DTMF tone to caller
ASTERISK-25950: [patch]SIP channel does not send PeerStatus events for autocreated peers
ASTERISK-25951: res_agi: run_agi eats frames it shouldn't
ASTERISK-25952: Hello ,I'm new AGI script in Asterisk and try to run small script that pass parameter from AGI to agi script and return result to dial plan but give me this error AGI Tx >> 510 Invalid or unknown command AGI Script ivrwl.agi completed, returning 0
ASTERISK-25953: Segfault in Queue
ASTERISK-25954: Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName
ASTERISK-25955: Make single-connection per dns for func_odbc optional
ASTERISK-25956: Compilation error in conditionally compiled code in config_options.c
ASTERISK-25957: Segfault in odbc after commit 9b0a96b947437f58fcc88f154ed5080fde529009
ASTERISK-25958: tests/app/mixmonitor: Sporadic failure due to incorrect size
ASTERISK-25959: http_media_cache/retrieve_cache_control_directives: Sporadic failure
ASTERISK-25960: The config_hook unit test causes Asterisk to crash if run a second time
ASTERISK-25961: tests/channels/SIP/sip_tls_call: Sporadic crash when running test
ASTERISK-25962: Asterisk crashes, potential cause: realtime musiconhold
ASTERISK-25963: func_odbc requires reconnect checks for stale connections
ASTERISK-25964: Outbound registrations created via ARI/push configuration do not clean up outbound registrations currently in flight
ASTERISK-25965: res_pjsip_outbound_publish: Allow multiple clients per configuration
ASTERISK-25966: Database based hints empty during AMI reload or startup
ASTERISK-25967: testsuite: Fix premature stopping of manager redirect dual tests
ASTERISK-25968: pjproject_bundled: Configure and make need to be re-tested
ASTERISK-25969: tests/channels/SIP/sip_bye_also: Sporadic failures
ASTERISK-25970: Segfault in pjsip_url_compare
ASTERISK-25971: WebRTC - set Asterisk IP for SDP manually
ASTERISK-25972: res_pjsip_exten_state: Use body generator to publish extension state
ASTERISK-25973: Asterisk crashes when call busy agent is enabled
ASTERISK-25974: Unused realtime MOH classes not purged on 'moh reload'
ASTERISK-25975: Asterisk 11.22.0 crashes due to error in app_queue.so
ASTERISK-25976: configs/basic-pbx/asterisk.conf contains incorrect path separator
ASTERISK-25977: network-manager uninstalled on ubuntu desktop 14.04
ASTERISK-25978: res_pjsip_authenticator_digest: Should not use source port in nonce verification
ASTERISK-25979: res_pjsip: Weird flood of traffic when authentication fails on TCP
ASTERISK-25980: [patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
ASTERISK-25981: ARI: PlaybackStopped shows old channel id after attended transfer
ASTERISK-25982: [patch]res_fax/t38_gateway: Peer V.21 session is created on wrong channel
ASTERISK-25983: testsuite: Need an install_prereq script
ASTERISK-25984: res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
ASTERISK-25985: chan_sip:sip_poke_peer does't copy port
ASTERISK-25986: chan_sip: Wrong request line with IMS and Proxy - Asterisk should be using loose routing, but it isn't?
ASTERISK-25987: testsuite: Correct PJSIP tag to pjsip on 5 attended transfer tests
ASTERISK-25988: Asterisk is lacking a systemd unit file
ASTERISK-25989: apps/confbridge: add regcontext feature
ASTERISK-25990: PJSIP TLS registration should respect client_uri scheme when generating Contact URI
ASTERISK-25991: ASAN: double free in res_odbc.c
ASTERISK-25992: How to get all the call status event in c#.net program
ASTERISK-25993: pjproject: Allow bundling to not require everything it does
ASTERISK-25994: [patch]res_pjsip: module load priority
ASTERISK-25995: dpma-firmware.json on downloads.digium.com inconsistent with filenames
ASTERISK-25996: Remove "live_dangerously" requirement on DB(read)
ASTERISK-25997: testsuite: Rest API tests that use autobahn fail with versions >= 0.13.1
ASTERISK-25998: file: Crash when using nativeformats
ASTERISK-25999: res_pjsip_dialog_info_body_generator: Remove subscription requirement